1 2 /* ----------------------------------------------------------------------------------------------------------- 3 Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5 Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. 6 All rights reserved. 7 8 1. INTRODUCTION 9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements 10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. 11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part 16 of the MPEG specifications. 17 18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) 19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners 20 individually for the purpose of encoding or decoding bit streams in products that are compliant with 21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license 22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec 23 software may already be covered under those patent licenses when it is used for those licensed purposes only. 24 25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, 26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional 27 applications information and documentation. 28 29 2. COPYRIGHT LICENSE 30 31 Redistribution and use in source and binary forms, with or without modification, are permitted without 32 payment of copyright license fees provided that you satisfy the following conditions: 33 34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or 35 your modifications thereto in source code form. 36 37 You must retain the complete text of this software license in the documentation and/or other materials 38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. 39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your 40 modifications thereto to recipients of copies in binary form. 41 42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without 43 prior written permission. 44 45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec 46 software or your modifications thereto. 47 48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software 49 and the date of any change. For modified versions of the FDK AAC Codec, the term 50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term 51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 52 53 3. NO PATENT LICENSE 54 55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, 56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with 57 respect to this software. 58 59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized 60 by appropriate patent licenses. 61 62 4. DISCLAIMER 63 64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors 65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties 66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, 68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits, 69 or business interruption, however caused and on any theory of liability, whether in contract, strict 70 liability, or tort (including negligence), arising in any way out of the use of this software, even if 71 advised of the possibility of such damage. 72 73 5. CONTACT INFORMATION 74 75 Fraunhofer Institute for Integrated Circuits IIS 76 Attention: Audio and Multimedia Departments - FDK AAC LL 77 Am Wolfsmantel 33 78 91058 Erlangen, Germany 79 80 www.iis.fraunhofer.de/amm 81 amm-info (at) iis.fraunhofer.de 82 ----------------------------------------------------------------------------------------------------------- */ 83 84 /***************************** MPEG Audio Encoder *************************** 85 86 Initial Authors: Markus Multrus 87 Contents/Description: PS Wrapper, Downmix header file 88 89 ******************************************************************************/ 90 91 #ifndef __INCLUDED_PS_MAIN_H 92 #define __INCLUDED_PS_MAIN_H 93 94 /* Includes ******************************************************************/ 95 #include "sbr_def.h" 96 #include "qmf.h" 97 #include "ps_encode.h" 98 #include "FDK_bitstream.h" 99 #include "FDK_hybrid.h" 100 101 102 /* Data Types ****************************************************************/ 103 typedef enum { 104 PSENC_STEREO_BANDS_INVALID = 0, 105 PSENC_STEREO_BANDS_10 = 10, 106 PSENC_STEREO_BANDS_20 = 20 107 108 } PSENC_STEREO_BANDS_CONFIG; 109 110 typedef enum { 111 PSENC_NENV_1 = 1, 112 PSENC_NENV_2 = 2, 113 PSENC_NENV_4 = 4, 114 PSENC_NENV_DEFAULT = PSENC_NENV_2, 115 PSENC_NENV_MAX = PSENC_NENV_4 116 117 } PSENC_NENV_CONFIG; 118 119 typedef struct { 120 UINT bitrateFrom; /* inclusive */ 121 UINT bitrateTo; /* exclusive */ 122 PSENC_STEREO_BANDS_CONFIG nStereoBands; 123 PSENC_NENV_CONFIG nEnvelopes; 124 LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ 125 126 } psTuningTable_t; 127 128 /* Function / Class Declarations *********************************************/ 129 130 typedef struct T_PARAMETRIC_STEREO { 131 HANDLE_PS_ENCODE hPsEncode; 132 PS_OUT psOut[2]; 133 134 FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2][MAX_HYBRID_BANDS]; 135 FIXP_DBL *pHybridData[HYBRID_READ_OFFSET+HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; 136 137 FIXP_QMF qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>1][QMF_CHANNELS]; 138 int qmfDelayScale; 139 140 INT psDelay; 141 UINT maxEnvelopes; 142 UCHAR dynBandScale[PS_MAX_BANDS]; 143 FIXP_DBL maxBandValue[PS_MAX_BANDS]; 144 SCHAR dmxScale; 145 INT initPS; 146 INT noQmfSlots; 147 INT noQmfBands; 148 149 FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2*HYBRID_FILTER_LENGTH*HYBRID_MAX_QMF_BANDS]; 150 FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2*HYBRID_FILTER_DELAY*(QMF_CHANNELS-HYBRID_MAX_QMF_BANDS)]; 151 FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; 152 FDK_SYN_HYB_FILTER fdkHybSynFilter; 153 154 } PARAMETRIC_STEREO; 155 156 157 typedef struct T_PSENC_CONFIG { 158 INT frameSize; 159 INT qmfFilterMode; 160 INT sbrPsDelay; 161 PSENC_STEREO_BANDS_CONFIG nStereoBands; 162 PSENC_NENV_CONFIG maxEnvelopes; 163 FIXP_DBL iidQuantErrorThreshold; 164 165 } PSENC_CONFIG, *HANDLE_PSENC_CONFIG; 166 167 typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; 168 169 170 /** 171 * \brief Create a parametric stereo encoder instance. 172 * 173 * \param phParametricStereo A pointer to a parametric stereo handle to be allocated. Initialized on return. 174 * 175 * \return 176 * - PSENC_OK, on succes. 177 * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. 178 */ 179 FDK_PSENC_ERROR PSEnc_Create( 180 HANDLE_PARAMETRIC_STEREO *phParametricStereo 181 ); 182 183 184 /** 185 * \brief Initialize a parametric stereo encoder instance. 186 * 187 * \param hParametricStereo Meta Data handle. 188 * \param hPsEncConfig Filled parametric stereo configuration structure. 189 * \param noQmfSlots Number of slots within one audio frame. 190 * \param noQmfBands Number of QMF bands. 191 * \param dynamic_RAM Pointer to preallocated workbuffer. 192 * 193 * \return 194 * - PSENC_OK, on succes. 195 * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. 196 */ 197 FDK_PSENC_ERROR PSEnc_Init( 198 HANDLE_PARAMETRIC_STEREO hParametricStereo, 199 const HANDLE_PSENC_CONFIG hPsEncConfig, 200 INT noQmfSlots, 201 INT noQmfBands 202 ,UCHAR *dynamic_RAM 203 ); 204 205 206 /** 207 * \brief Destroy parametric stereo encoder instance. 208 * 209 * Deallocate instance and free whole memory. 210 * 211 * \param phParametricStereo Pointer to the parametric stereo handle to be deallocated. 212 * 213 * \return 214 * - PSENC_OK, on succes. 215 * - PSENC_INVALID_HANDLE, on failure. 216 */ 217 FDK_PSENC_ERROR PSEnc_Destroy( 218 HANDLE_PARAMETRIC_STEREO *phParametricStereo 219 ); 220 221 222 /** 223 * \brief Apply parametric stereo processing. 224 * 225 * \param hParametricStereo Meta Data handle. 226 * \param samples Pointer to 2 channel audio input signal. 227 * \param timeInStride, Stride factor of input buffer. 228 * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. 229 * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. 230 * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. 231 * \param downsampledOutSignal Pointer to buffer where to write downmixed timesignal. 232 * \param sbrSynthQmf Pointer to QMF synthesis filterbank. 233 * \param qmfScale Return scaling factor of the qmf data. 234 * \param sendHeader Signal whether to write header data. 235 * 236 * \return 237 * - PSENC_OK, on succes. 238 * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. 239 */ 240 FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( 241 HANDLE_PARAMETRIC_STEREO hParametricStereo, 242 INT_PCM *samples[2], 243 UINT timeInStride, 244 QMF_FILTER_BANK **hQmfAnalysis, 245 FIXP_QMF **RESTRICT downmixedRealQmfData, 246 FIXP_QMF **RESTRICT downmixedImagQmfData, 247 INT_PCM *downsampledOutSignal, 248 HANDLE_QMF_FILTER_BANK sbrSynthQmf, 249 SCHAR *qmfScale, 250 const int sendHeader 251 ); 252 253 254 /** 255 * \brief Write parametric stereo bitstream. 256 * 257 * Write ps_data() element to bitstream and return number of written bits. 258 * Returns number of written bits only, if hBitstream == NULL. 259 * 260 * \param hParametricStereo Meta Data handle. 261 * \param hBitstream Bitstream buffer handle. 262 * 263 * \return 264 * - number of written bits. 265 */ 266 INT FDKsbrEnc_PSEnc_WritePSData( 267 HANDLE_PARAMETRIC_STEREO hParametricStereo, 268 HANDLE_FDK_BITSTREAM hBitstream 269 ); 270 271 #endif /* __INCLUDED_PS_MAIN_H */ 272
